March 2025 - sr-users - kamailio.org

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 17 Apr '26

17 Apr '26

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

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dynamic selection of authentication algorithm
by Juha Heinanen 28 Oct '25

28 Oct '25

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

5 13

SIP Trunk registration In Kamailio
by tanmoykarmakar10＠gmail.com 14 Oct '25

14 Oct '25

I want to register SIP Trunk directly in Kamailio. How can I register the SIP trunk in Kamailio. Please find the below sample SIP trunk details. Network IP 10.2.2.2 Subnet Mask 255.255.255.252 Customer IP 10.3.3.3 Gateway IP 10.1.1.1 SBCIP 10.4.4.4 Username 12345678 Password 1234 Please help me with configuration to register SIP trunk in kamailio.

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Support for TLS server_name extension (aka SNI=server name indication)
by Barry Flanagan 04 Sep '25

04 Sep '25

Hi, Back in Kamailio 1.5, the release notes state: "support for TLS server_name extension (aka SNI=server name indication)" However, I cannot find any indication of this in the current TLS docs, and trying to set tls_server_name or server_name in tls.cfg fails with "unsupported option". Is this actually supported? Thanks. -Barry Flanagan

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[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 02 Sep '25

02 Sep '25

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

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[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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How to place a delay between cancelling branch and retrying next one on timer expiry?
by Alex A 15 Apr '25

15 Apr '25

Hi All, I have an interesting use-case where I need to place a delay between cancelling branch and retrying next one. *Background: * Running Kamailio 5.6.x Kamailio routes requests via a freeswitch B2BUA server on select call flows (where B2BUA functionality is required). Call flow examples: PSTN --> Kamailio --> Freeswitch --> appserver1 PSTN --> Kamailio --> Freeswitch --> appserver2 On attempt failures, a branch retry is required from appserver1 to appserver2; however freeswitch B2BUA remains the same Since Freeswitch does not quite follow the RFC 3261 timer K specifications (should be 0 seconds for reliable transports, but FS keeps transactions for T4 (set to 2 seconds)); I have implemented an artificial delay in failure route before retrying (as per recommendation described here: https://freeswitch-users.freeswitch.narkive.com/TsSye66D/482-request-merged… ) *Script config: * onreply_route[MANAGE_REPLY] { xlog("L_INFO", "reply $T_reply_code received"); if (t_check_status( "1[1-9][0-9]")) { t_reset_fr(); } else { t_set_fr(120000); } } route(NATDETECT); #!ifdef WITH_RTPENGINE route(RTPMANAGE); #!endif } route[RELAY] { if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); } if (!t_relay()) { sl_reply_error(); } exit; } route[TOCARRIER] { process bunch of things, attach custom headers, generate request URI t_on_failure("MANAGE_FAILURE_CARRIER"); route(RELAY); } failure_route[MANAGE_FAILURE_CARRIER] { xlog("L_INFO", "reply $T_reply_code received"); if ($avp(inbound_attempt)<=1){ if ($dlg_var(b2bua)=="true"){ usleep(1900000); } route("TOCARRIER"); } *Call Flows: * The above fix works well when an explicit error reply is received back. *error reply SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 503 <-- Freeswitch failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch call continues as expected. however does not work when NO reply is received at all and the timer expires. *Current timer expiry SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 100/180 <-- Freeswitch SBC (timer expires) failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch SBC <-- 482 Merged <-- Freeswitch SBC --> CANCEL(branch0) ---> Freeswitch I am looking to CANCEL branch0 before the sleep delay runs Based on source code, t_cancel_branches("this") can only be executed in reply_route; however the chellenge is that reply_route never sees the "faked 408" on timer expiry(although the failure route does see $T_reply_code 408). Any suggestions on how to handle this would be greatly appreciated. TIA

6 11

DMQ read message too large
by Calvin E. 31 Mar '25

31 Mar '25

We are seeing errors like this when synchronizing htables over DMQ: ERROR: <core> [core/udp_server.c:677]: udp_rcv_loop(): read message too large: 60793 We have no issues with other UDP fragments caused by STIR/SHAKEN headers. Our socket buffers are sufficiently large (SO_RCVBUF is initially 26214400). Does DMQ have a receive message size limit less than what it can send? Is DMQ intended for htables under a certain total size? We tried to configure DMQ to use a TCP listener with ;transport=tcp on the notification and server address, but the DMQ traffic was still sent as UDP. The example in the documentation shows TLS, but we haven't tried that. -Calvin E.

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Intermediate CDRs
by Sergio Charrua 31 Mar '25

31 Mar '25

Hi all! I have been googling, and reading documentation, but I could not find any solution. How can I implement intermediate CDRs with Kamailio ACC module (or any other module) ? For example, on long duration calls, I 'm required to save call status on a CDR, resulting in intermediate CDR lines for the call (which will result in multiple CDR lines per long duration calls). Any clue on how to implement this? Did anyone achieved that with Kamailio ACC module? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

4 3

Multiple code modules
by David Cunningham 28 Mar '25

28 Mar '25

Hello, We have a requirement to load two code modules for use by kamailio.cfg, the reason being that we can't (won't, for business reasons) modify the first module and so need to add a second for our own code. Ideally we would load two Perl modules instead of the single module that's currently loaded like this: modparam( "app_perl", "filename", "/path/to/our/custom/Kamailio.pm" ) But I'm not sure how to load two modules like this. Otherwise, does anyone know if it's safe to load both a Perl module and a Python module? Then we could at least implement both required parts, one in Perl and one in Python. Thanks in advance for any help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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